Route audio from external SIP client into Asterisk

Asterisk installation with SIP clients on LAN. Asterisk running on local Linux server. Also running on the same server is a ‘SIP communicator’ basically a soft SIP client that registers to an external SIP server i.e. not the local Asterisk server. This client is controlled via UDP messages from another application and communicates with other clients on the external SIP server using IM with optional audio.

Need to be able to route audio from SIP communicator to/from a SIP client registered on the local Asterisk server. What is the best way to achieve this?

I don’t think there is enough information to answer this. Also, are you sure you only want to handle outgoing calls, otherwise your SIP clients become SIP servers.

In particular, how can you communicate signalling to SIP communicator, and how can you communicate media to it, except using SIP (which would make this too trivial for you to need to ask).

Hi, thanks for your response.Maybe I didn’t describe the scenario very well. It’s also quite possible that the solution is trivial and lack of knowledge is my problem…

The SIP communicator is standalone, registers only to the external SIP server, and is not linked to Asterisk in any way. Audio in/out is normally via audio hardware on the server i.e. analogue. I’m trying to find a way that the SIP communicator audio can be accessed at the SIP level and connected into Asterisk. I could have another soft client on a different box - this time registered to Asterisk - and patch the audio at the analogue level. However I’m thinking there must be a way to do this at the SIP level?

SIP is SIP. If a call is made and SDP negotiated with RTP, then media will flow. Where that media is connected to within the client is up to it, and same for Asterisk.

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